This invention relates, in general, to an adaptable interface between a broadband-narrowband network and in which adaptation is required between time division multiplexed (TDM) communication and asynchronous transmission mode (ATM) communication. More particularly, but not exclusively, the present invention is applicable to an interface provisioned to support numerous different adaptation protocols, especially adaptation layer two (AAL-2), and a method of packetizing information to optimise utilisation of available bandwidth.
Globally, telecommunication systems can generally be considered to be in a transitional phase between first generation narrowband digital networks (such as the global system for mobile (GSM) cellular communication system) and future multi-media digital networks (such as the universal mobile telecommunication system (UMTS)) having broadband capabilities. Indeed, radio frequency (RF) and wireline systems are being merged together to enhance the information transfer mechanism, while still providing some flexibility with respect to mobility within the network. For example, broadband (typically fibre-optic based) infrastructure connections are being utilised to support information (both voice and data) transfer between cellular RF coverage areas. The transition to broadband systems is, in fact, necessarily required to support higher data rate communications, including video and Internet applications that are presently being both considered and made available to subscribers to the service.
A key goal of development of telecommunications networks is to realise the potential integration of real-time and non-real-time services. The key examples of these two types are voice telephony and computer data. Voice telephony is served predominantly by a circuit switched connection orientated network, arranged to deliver a guaranteed quality of service (QoS). Such networks are implemented by transport and switching systems that use a time division multiplexing scheme. Computer data is served predominantly by the Internet that uses a packet forwarding connectionless mode of operation, i.e. a workable paradigm best suited to the burstiness of traffic demand and general non-deterministic nature of this traffic type.
The technologies and protocols that will serve the integration of these two different types of service are presently being decided by network operators. In this respect, the two main contenders for universal transport and switching are ATM and Internet Protocol (IP), although there are many other legacy systems and nascent technologies that may offer specialised solutions to carrying key services.
Present broadband digital networks are characterised in that user and control information is transmitted in fixed xe2x80x9cpacketxe2x80x9d lengths for the duration of a call, with these packets pre-pended with headers that contain bearer channel identification. Such a broadband system is described in the requirement of the ATM Forum Utopia Level 1/2 Interface. In contrast with narrowband systems, user information is relayed across a node via an asynchronous switching fabric that examines each packet in turn (using some kind of fairness algorithm) and directs it to the appropriate output link in response to the input link and bearer channel identification. Routing and control information transmissions are, however, similar to that for the narrowband case and differ only in as much as the signalling schemes are technology specific.
To facilitate use of broadband networks and the migration of communication networks to high data-rate technologies (e.g. the two mega-bit per second rate envisaged within UMTS), there is a need to provide an effective mechanism for interconnecting narrowband networks through a transparent broadband ether. In other words, the broadband ether must accommodate and support narrowband signalling schemes without affecting either data integrity or in any way inhibiting data flow or interconnection. As such, a narrowband-broadband interface must contain adaptation modules that freely translate between TDM and ATM, for example.
ATM has been designed from the outset to adapt to many different types of communications traffic. ATM is a connection orientated network mechanism, allowing dynamic bandwidth configuration and flexibility as a key advantage over circuit switched networks. ATM has adaptation layers for carrying given services over ATM transport and switches. However the fixed length of ATM cells, while suitable for segmentation of long data packets, thereby simplifying and streamlining switching technologies, is still too large for certain compressed voice services, that suffer a xe2x80x98cellificationxe2x80x99 delay, affecting existing network delay budgets and acceptable voice characteristics. Indeed, when considering the issues of delay, meaningful voice communication across a channel is achieved with a pure delay of less than one hundred and fifty (150) milliseconds. However, since the signal is likely to suffer from echo, telecommunications standard bodies have stipulated that echo cancellation must be applied to all channels having a one-way delay of greater than twenty-five milliseconds. Unfortunately, in relation to cell assembly of a sixty-four kbps PCM voice channel, a delay of six milliseconds is introduced merely by the provision of sampling. Consequently, an allowable path delay is immediately reduced to nineteen milliseconds, which reduced period is easily exceeded in moderate and long distance calls, e.g. a long distance call between Washington D.C. and San Francisco. Furthermore, conversion of the cell into a narrowband component for onward routing of the call further reduces the available nineteen milliseconds assigned to accommodate all additional delays.
There are, in fact, already a plethora of broadband adaptation schemes that are presently employed or which are being developed or evolved to cope with broadband transmissions. Specifically, ATM adaptation layer protocols such as AAL-1 (and structured data transfer, SDT), AAL-2 and AAL-5 impose very different requirements on processing capabilities of a communication network, especially in relation to a narrowband-broadband interface.
AAL-1 is an ATM adaptation protocol targeted at constant bit rate (CBR) traffic, e.g. voice or video, and is applicable to data rates equal to or exceeding sixty-four kbps (64 kbps). More particularly, narrowband voice signals at 64 kbps are packaged into ATM cells having an overall length of fifty-three bytes; five bytes of which are used as a cell header whilst the remaining forty-eight bytes support the payload. The cell header contains control and routing information, such as the virtual circuit identifier (VCI). With respect to the payload in AAL-1 SDT (structured data trasfer), a first byte (or xe2x80x9coctetxe2x80x9d) is reserved for a sequence number that provides an error correction facility, while the remaining forty-seven octets are allocated to voice samples. Every eight cells, the first byte is stolen from the forty-seven octets allocated to voice samples of that cell, and is stolen byte is used as a pointer to indicate a structure boundary. The pointer field therefore allows multiplexed transmissions of multiple voice channels, as will be appreciated.
ATM cells can also be formatted in 16-bit words, rather than the more frequently used eight bit word structures described in the immediately preceding paragraph. To accommodate this increase in traffic, each ATM cell is extended to fifty-four bytes, arranged in words with the first three words used for the ATM cell header and twenty-four words sent as traffic in the remaining words of the extended ATM cell.
Eight and sixteen bit buses are particularly suited to byte-based services, and also give backwards compatibility with the ATM Forum Utopia Level 1/2 Interface (see The ATM Forum, xe2x80x9cUtopia, AN ATM-PHY Interface Specification Level 1, Version 2.01xe2x80x9d, Mar. 21st 1994 (AF-PHY-0017.000), and The ATM Forum Utopia Level 2. Version 1.0, June 1995 (AF-PHY-0039.000)).
In the case of both the fifty-three byte and fifty-four byte ATM cells, the transmission length of the ATM cell is fixed and constant for the entire duration of the call.
AAL-5 provides a capability of data and voice transmissions to work stations, and is therefore particularly applicable to multi-media communication systems. AAL-5 segments long data structures into many cells, with a data structure conceivably exceeding fifteen hundred octets in length. AAL-5 similarly provides error correction in the packetised header, while also using a bit in the header to indicate the continuation or end of a long data structure. Furthermore, control information included within the AAL-5 protocol also stipulates how may cells have been consolidated together to produce the long data structure and will also include CRC check bit information for error correction. When carrying voice, the AAL-5 structure is typically only one cell in length and, as such, may require the provision of echo cancellers. However, the support of data communications by AAL-5 is inherently immune to delay because data transmissions are not time dependent for coherent understanding.
ATM Adaptation Layer 2 (AAL-2) is distinct from other ATM adaptation layers since it de-couples voice packets from ATM cell boundaries, and also since mini-packets from several calls can be multiplexed into a single ATM connection. This multiplex is asynchronous to the cell boundary and further effectively introduces a new switching layer above the ATM layer.
More specifically, the AAL-2 adaptation scheme is designed to support compressed voice at or below rates of sixty-four kbps. Indeed, compression algorithms, such as ADPCM (adaptive differential pulse code modulation) or LD-CELP (low-delay, code excited linear prediction) can enable voice channels to be compressed from sixty-four kbps to rates typically between four kbps and thirty-two kbps. Additional channel utilisation can be achieved by suppressing any silent intervals that occur naturally in speech (especially bearing in mind that one person in a call is typically silent when listening to the other party talk), which suppression can enable a further bandwidth saving of over 50%. As will be appreciated, the communication system is most efficient when maximum traffic is being passed, i.e. at a rate of 64 kbps, although lower rates utilising coding techniques are subject to flow control constraints to maintain cell timing/arrival (i.e. cell flow) and cell delination.
AAL-2 services may be packet based as well as TDM based.
With respect to AAL-1, it is not possible to support silence suppression and it generally takes too long to fill an ATM cell with low data rate compression schemes. In fact, in an eight kbps compression scheme, it takes forty-seven milliseconds to fill an ATM cell and this results in an unacceptable delay (and the necessity for the use of expensive echo-cancellation techniques) in voice communication. These problems, however, can be addressed and to some extent resolved by using AAL-2 that multiplexes a number of compressed voice channels into a single ATM virtual channel (VC). In fact, the packet lengths in AAL-2 can be arbitrarily small (from one octet to sixty-four octets in length) which enables the packet size (and hence the sampling delay) to be tailored to the compression rate being used (e.g. at 8 kbps a packet payload size of eight octets takes just eight milliseconds to fill).
Looking at byte insertion rates, a 64 kbps codec generates eight thousand (8000) bytes per second or one byte every one hundred and twenty-five microseconds (125 xcexcs). In contrast, a 32 kbps codec produces 4000 bytes per second or one byte every 250 xcexcs. In fact, AAL-2 actually partitions data into forty byte packets (equivalent to five millisecond (5 ms) time segments).
AAL-2 allows up to two-hundred and fifty-six mini-channels to be supported on a single ATM virtual channel (VC). Furthermore, as will be understood, AAL-2 is a relatively new standard that, at present, is not widely used and, in some instances, is less efficient than AAL-1. More specifically, in the latter respect and specifically in relation to relatively long data structures, AAL-2 requires an increased control overhead associated with the provision of additional mini-cell headers.
Clearly, in the event that a transmitter has optional selection of codec rate, e.g. based on line quality, it is essential that different packet (and hence rates) are clearly associated to avoid misinterpretation of data caused by incorrect timing recovery.
In an attempt to improve efficiency in terms of packaging information into ATM cells, it has previously been contemplated that the cell length could be extended or reduced by clearly marking the end of a cell. However, this idea is not physically compatible with the existing Utopia interface standard for ATM transmission. More especially, the marking of the end of packet locations requires the use of an end of packet identifier and hence a distinct and separate data line into an ATM interface. Putting this another way, the previous proposal could not include end of packet information inherently within the combined data, address and control bus originating from the TDM domain. Consequently, the combined bus would therefore need to be physically widened, i.e. extended, to support the inclusion of a further control bit associated with the subsequent inclusion within the ATM cell of an end of cell marker.
To complete the picture, Internet Protocol (IP) was designed for computer communications, although it has been recently demonstrated to be suitable for real-time services if congestion can be controlled to an extent that permits an acceptable quality of service to be achieved. Much activity surrounds investigations and implementation of mechanisms for limiting the degree of congestion and controlling quality of service in IP networks, including the Real-Time Protocol (RTP) that was invented to carry realtime traffic, in particular voice and video services that the existing transport protocols (TCP and UDP) cannot cater for. RTP takes into account the lossy behaviour of IP networks when congested, in a manner suitable to real-time services. UDP has no necessary timing or sequence information; TCP does not account for the low-delay and immediacy of real-time service requirements, e.g. a late packet should be dropped rather than retransmitted.
In IP, congestion and packet loss can be limited by reserving bandwidth on key routes in the IP network, as implemented in the Reservation Protocol (RSVP). This mechanism in effect makes the connectionless network behave as if connection orientated, since all packets entitled to use the reserved bandwidth must follow an established route. Such IP mechanisms are generally analogous to ATM QoS and virtual path (VP) and virtual connection (VC) partitioning. However, the particular issue of IP packetisation delay affecting delay sensitive services has not been addressed for two reasons, namely IP packets may be variable length (and therefore the packetisation delay is arbitrary); and IP networks carrying voice are, in practice, yet to achieve an acceptable QoS.
Therefore, as will be appreciated, because of the limited availability of bandwidth (arising from limited spectrum, increasing numbers of users and enhanced services) in any communication system, including broadband ATM systems, it is always desirable to optimise transmission formats to enhance data throughput. Furthermore, subscribers demand an acceptable quality of service, especially in relation to voice communication, which quality of service cannot presently be provided by IP techniques. A need has therefore been created for an interface to an adaptation device, with characteristics that are flexible enough in particular to take full advantage of the enhanced features of AAL-2, whilst providing compatibility with other adaptation layers.
According to a first aspect of the present invention there is provided a method of sending information in packets over a multiplicity of frames, the method comprising: pre-partitioning each frame into a plurality of slots of which at least some may have differing durations to others, each of the plurality of slots associated with at least a defined packet length and a channel; and packetising information into a slot allocated to a call for transmission to an addressed unit, the packetised information containing a truncated header of reduced length arising from the slot allocated to the call inherently identifying the packet length thereof and the channel.
In another aspect of the present invention there is provided a packet interface for coupling between a narrowband domain and a broadband domain, the packet interface arranged to packetise information from the narrowband domain to the broadband domain into variable length packets, the packet interface having associated therewith: a processor arranged to construct a plurality of contiguous frames with each frame pre-partitioned into a plurality of slots of which at least some may have differing durations to others, each of the plurality of slots associated with at least a defined packet length and a channel, the processor further comprising: means to packetise information from the narrwoband domain into a slot selectively allocated to a call for transmission to an addressed unit, the packetised information containing a truncated header of reduced length arising from the slot allocated to the call inherently identifying the packet length thereof and the channel.
Preferably, the processor further comprises: means to assess at least one of a data transmission requirement for the call and a subscriber processing capability of the addressed unit in the call; and means to select and allocate the slot within the frame based on the assessment
The processor may further comprise means for notifying the addressed unit of the slot allocated to the call.
In another aspect of the present invention there is provided a method of sending information in packets comprising: receiving information from a channel and constructing at least one intermediate packet associated with that channel during a first time segment; generating an intermediate packet header having an intermediate packet length indication identifying a number of intermediate bytes received from the channel within the first time segment; communicating the intermediate packet header and the at least one intermediate packet to a broadband signal processor; receiving the intermediate packet header at the broadband signal processor and interrogating the intermediate packet header to identify the number of intermediate bytes that can be expected at the broadband signal processor during a second time segment; determining whether the at least one intermediate packet to be received at the broadband signal processor should be temporarily stored or converted into a broadband packet for onward transmission; and generating a broadband packet including a header containing a length indication if the at least one intermediate packet is to be converted into a broadband packet for substantially instantaneous onward transmission, else temporarily storing the at least one intermediate packet and incrementing a record of a number intermediate packets temporarily stored in relation to said channel.
The length indication in the broadband packet is variable and dependent upon the number of intermediate packets that are to be transmitted in the broadband packet.
The method preferably further comprises repeatedly storing contiguous intermediate packets incident on the broadband signal processor during successive time segments and incrementing the record of the number intermediate packets temporarily stored in relation to said channel.
The method may comprise comparing the record of the number of intermediate packets with a predetermined threshold and constructing a broadband packet and associated length indication when the record of the number of intermediate packets is at least equal to the predetermined threshold.
A particular embodiment of includes the step of temporarily storing the intermediate packets is overridden on the basis of one of an addressing unit""s identity and an addressed unit""s identity.
In yet another aspect of the present invention there is provided a packet interface comprising: a) a first buffer and processor combination arranged to receive information from a channel and to construct intermediate packets associated with that channel during a first time segment, the processor including: means for generating an intermediate packet header having an intermediate packet length indication identifying a number of intermediate bytes received from the channel within the first time segment; and means for communicating the intermediate packet header and the intermediate packets to a broadband signal processor; b) a signal processor and channel buffer combination, the signal processor coupled to receive the intermediate packet header and arranged to interrogate the intermediate packet header to identify the number of intermediate bytes that can be expected at the broadband signal processor during a second time segment, the signal processor further comprising: means for determining whether the intermediate packet to be received should be temporarily stored or converted into a broadband packet for onward transmission; and means for generating a broadband packet including a header containing a length indication if an incident intermediate packet is to be converted into a broadband packet for substantially instantaneous onward transmission, else means for temporarily storing the incident intermediate packet in the channel buffer and incrementing a record of a number intermediate packets temporarily stored in relation to said channel.
The length indication in the broadband packet may be variable and dependent upon the number of intermediate packets that are to be transmitted in the broadband packet.
In a particular embodiment, the signal processor and channel buffer are further arranged to repeatedly store contiguous intermediate packets incident on the broadband signal processor during successive time segments and wherein the signal processor further includes means for incrementing the record of the number intermediate packets temporarily stored in relation to said channel.
The signal processor preferably further comprises: means for comparing the record of the number of intermediate packets with a predetermined threshold; and means for constructing a broadband packet and associated length indication when the record of the number of intermediate packets is at least equal to the predetermined threshold.
The signal processor may be arranged to override the temporary storage of the intermediate packets on the basis of one of an addressing unit""s identity and an addressed unit""s identity.
The packet interface is typically included within a broadband/narrowband communication network, and especially in relation to a TDM/ATM interworking environment.
The preferred embodiments of the present invention can therefore support a dynamic change in packet length (i.e. the use of variable packet lengths in a broadband environment) in each connection during a call, with the system of the preferred embodiment able to provide multiple channel capacity to supply a switching adaptation device capable of supporting both a full range of voice services as well as data services. More particularly, the present invention is able to make better use of limited bandwidth resources by optimising data (i.e. payload and packet) transmission at the expense of header information considered redundant in relation to a specific call, i.e. there is no waste of bandwidth through optimised header encoding.
Beneficially, the preferred embodiments of the present invention provide a parallel signal interface for variable length packet transfer between hardware devices carrying a multiplex of connections. Indeed, the interface provides the means for full flow control on a packet by packet basis, where the packet length may be known implicitly in advance of transfer, known implicitly or explicitly to be limited in length in advance of transfer, or unknown implicitly or explicitly in advance of transfer. Furthermore, the interface can provide a mechanism for arbitration between multiple transmitters of packets and multiple receivers of packets and can also align a concatenation of shorter, fixed length packets to the start of longer packets for transmission and compatibility purposes.